A telephony operator may operate both a mobile network such as a Global System for Mobile communications (GSM) or a Universal Mobile Telecommunications System (UMTS) network on the one hand and a fixed line network such as Public Switched Telephone Network (PSTN) and Integrated Services Digital Network (ISDN) on the other hand.
The central component of the Core Network subsystem of a GSM network is the Mobile services Switching Center (MSC). It acts like a switching node of the PSTN or ISDN, and additionally provides functionality needed to handle a mobile subscriber, such as registration, authentication, location updating, handover, and call routing to a roaming subscriber. These services are provided in conjunction with several functional entities, which together form the Core Network subsystem. A Gateway MSC (GMSC) provides the connection to the fixed networks. Signaling between functional entities in the Core Network subsystem uses Signaling System Number 7 (SS7), used for trunk signaling in ISDN and widely used in current public networks.
The Home Location Register (HLR) together with the MSC, provide the call-routing and roaming capabilities of GSM. The HLR contains administrative information of each subscriber registered in the GSM network. The HLR also contains the physical address of the MSC/VLR where the mobile station is currently registered. The location of the mobile station is typically in the form of the signaling address of the VLR associated with the mobile station. The VLR contains selected administrative information from the HLR, necessary for call control and provision of the subscribed services, for each mobile station currently located in the geographical area controlled by the VLR and currently served by that VLR. Although each functional entity can be implemented as an independent unit, currently all manufacturers of switching equipment implement the VLR together with the MSC, so that the geographical area controlled by the MSC corresponds to that controlled by the VLR, thus simplifying the signaling required.
The administrative information of each subscriber comprises amongst others the International Mobile Subscriber Identity (IMSI), which is an internal subscriber identity used by the network, and a Mobile Station ISDN (MSISDN) number, which is the telephone number associated with the phone user.
There is logically one HLR per GSM network, although it may be implemented as a distributed database.
When a subscriber places a call to a mobile phone in the GSM network, he dials the MSISDN number, and the call is routed to the mobile phone operator's GMSC. A gateway is a node used to interconnect two networks. The gateway is often implemented in an MSC, in which case the MSC is referred to as the GMSC. The GMSC acts as the “entrance” from exterior portions of the Public Switched Telephone Network onto the provider's network.
As noted above, the phone user is free to roam anywhere in the operator's network or on the networks of roaming partners, including in other countries. The GMSC determines the MSC/VLR to which the mobile phone is registered in order to connect the call. It does this by consulting the HLR. The HLR relays a so called Mobile Station Roaming Number (MSRN) to the GMSC, which uses it to route the call to a Visited MSC (VMSC). The above described network architecture is both applicable to GSM networks, which are referred to as second generation (2G) mobile systems, and third generation (3G) mobile systems, which will follow up the 2G systems.
UMTS is one of these 3G mobile phone technologies. It uses Wideband Code Division Multiple Access(W-CDMA) as the underlying standard. UMTS is standardized by the 3GPP (3rd Generation Partnership Project), and represents the European/Japanese answer to the ITU IMT-2000 requirements for 3G mobile systems.
Currently, most fixed line networks and mobile networks use Circuit Switched (CS) technology for the media transport. There is, however, a gradual shift from CS based telecommunications networks to Packet Switched (PS) based telecommunications networks. PS based networks may use e.g. the Internet Protocol (IP) as protocol for signaling between network entities and for the transport of multimedia streams.
The IP Multimedia Subsystem (IMS) is a standardized networking architecture for telecommunication operators that want to provide mobile and fixed multimedia services. It uses a Voice-over-IP (VoIP) implementation based on a 3GPP standardized implementation of Session Initiation Protocol (SIP), and runs over the standard Internet Protocol (IP).
SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include e.g. voice calls, multimedia sessions, and multimedia conferences. SIP invitations (SIP INVITES) are used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.
IMS provides for a new telecommunication service delivery platform for operation with the Internet. With IMS, users have to be able to execute all their services from their home networks as well as when roaming from their home network to a visited network. To achieve these goals, IMS is based on standard IP protocols as defined by the IETF. So, a multimedia session between two IMS users, between an IMS user and a user on the Internet, and between two users on the Internet is established using the same protocol. Moreover, the interfaces for service developers are also based on IP protocols. Existing phone systems (both PS and CS) are supported by IMS and IMS may be applied in fixed networks and in mobile networks. Therefore IMS merges the Internet with the cellular world; it uses cellular technologies to provide ubiquitous access and Internet technologies to provide appealing services.
An operator that owns both a fixed line network and a mobile network may migrate the fixed line network from CS technology to IMS, prior to migrating the mobile network from CS technology to IMS. This trend, whereby IMS will be more widespread in fixed line networks than in mobile networks, is due to the fact that mobile networks typically do not have sufficient capacity in the Radio Access Network (RAN) to handle IMS for speech calls. Insufficient capacity may comprise too high latency or too low bandwidth. Hence, IMS will experience a slower take-up in the mobile network than in the fixed line network.
Such operator that operates both a fixed line network, needs efficient mechanisms to route calls between the fixed line network and the mobile network. The operator may offer services like Virtual Private Network (VPN), where subscribers have a fixed line connection. Within the context of VPN, calls need to be routed to e.g. from the fixed network to the mobile network. A fixed line connection may be through an ISDN/PSTN connection or a Private Branch eXchange. A Private Brach eXchange (also called PABX, PBX) is a telephone exchange that is owned by a private business, as opposed to one owned by a common carrier or by a telephone company.
FIG. 1 gives a graphical representation of call routing between the fixed line network and the mobile network. A fixed line call, initiated from within the PSTN, may need to be routed to the PLMN, under the control of the operator's VPN service. When the call is routed to the PLMN, the call is handled by the GMSC and the HLR according to the standard GSM/UMTS call handling process. The call is routed to the VMSC where the destination subscriber is currently registered. The call setup between Local Exchange (LE), GMSC and VMSC is done using ISDN User Part (ISUP). ISUP is part of the Signaling System #7 (SS7) which is used to set up telephone calls in PSTN. The
Bearer Independent Call Control (BICC, “layered architecture”) may be used instead of ISUP. When such operator has migrated the fixed line network to IMS, but not yet the mobile network, that operator still requires efficient mechanisms to route calls from IMS in the PSTN to PLMN. In such scenario, the call control and the media transport in the fixed line network are both based on IP. When the call is routed from PSTN to PLMN, the IP signaling and media transport is converted to the signaling and transport protocols used in the PLMN. This may be, as in the earlier example, ISUP/BICC and Time Division Multiplex (TDM) respectively.
FIG. 2 depicts a possible architecture for above-described scenario of Routing from an IP domain to a CS domain. The media stream of the call in the (fixed line) IP network is transported over Real-time Transport Protocol (RTP), whereas the signaling is handled by SIP. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP is independent of the underlying transport and network layers. The IMS related entities between the IMS terminal and the SIP Application Server (AS), such as Serving Call Session Control Function (S-CSCF), are not depicted in FIG. 2.
To route the call to the PLMN, a border Gateway is needed: the SIP signaling needs to be converted to ISUP and the RTP media stream needs to be converted to TDM. After conversion of the respective protocols, the call can be forwarded to the GMSC in the PLMN and from there to the VMSC.
In the following a current method is described for call routing via GMSC in PLMN. The MSISDN of the destination subscriber identifies that subscriber, but does not reveal her location. For that reason, the IMS network routes the call to that MSISDN using a Breakout Gateway Control Function (BGCF). The BGCF may be a default BGCF. The BGCF is located in the IMS network. The BGCF forwards the call to a GMSC in the GSM network of the destination subscriber. The GMSC takes care of routing the call further; this includes, amongst others, obtaining a Mobile Station Roaming Number (MSRN) from HLR, and routing the call to the appropriate MSC. The routing of the call from GMSC to VMSC is done with ISUP or BICC (for signaling) and TDM (or other protocol) for media stream.
A current development is the support of SIP in (G)MSC, for the purpose of routing calls in the PLMN for connecting to other networks.
In UMTS networks the MSC server controls all calls from a PSTN/ISDN/GSM network. The MSC server only takes care of the call control and service part, while the switch is replaced with a Media Gateway (MG). The MGs have different tasks depending on where they are placed in the network. The MG at the UMTS terrestrial radio access network (UTRAN) side transforms e.g. VoIP packets into radio frames and the MG at the PSTN side translates all calls coming from PSTN into VoIP calls for transport in a UMTS core network. The control of the MGs is managed by the Media Gateway Control Function (MGCF). The MGCF also performs translation at the call control signaling level between ISUP signaling, used in the PSTN and SIP signaling used in the UMTS multimedia domain. The Call State Control Function (CSCF) is a SIP server that provides/controls multimedia services for IMS mobile stations. The Signaling Gateway (SGW) just relays all the call-related signaling between UTRAN and PSTN on an IP bearer and sends the signaling data to the MGCF.
A prior art patent application WO 01/22766 describes a method for routing a call originated by a PSTN phone (PSTN) to a mobile station (MS) disposed in an integrated telecommunications network with a packet switched network (PSN) portion and a wireless circuit switched network (SN) portion including a location server (LS) containing mapping information between number information and Internet Protocol (IP) address information.